The present invention relates to Internet Telephony Signal Conversion.
Although originally intended for the transmission of computer data, more recently the Internet has been exploited to provide real time telephony communications. The primary attraction of the Internet for telephony communications is the low charge compared with conventional telephony. Many Internet users have a dial-up connection to an access provider over a local telephone line, and therefore such users pay only local telephone charges when logged on. Some access providers charge a monthly description, whilst others charge on the basis of connection time (some may do both). However, there is generally no charge associated with actual data transfer over the network. As a result, the effective cost of an international call over the Internet may be no more than that of a local call of the same duration to the access provider. In addition, the fully digital nature of the Internet may potentially offer a richer functionality (eg in terms of conference calling) than conventional telephone networks. Internet phones are surveyed in the article "Dial 1-800-Internet" in Byte Magazine, February 1996, pages 83-88 and in the article "Nattering On", in New Scientist, Mar. 2, 1996, pages 38-40.
The transmission of voice signals over a packet network is described for example in "Using Local Area Networks for Carrying Online Voice" by D. Cohen, pages 13-21, in "Voice Transmission over an Ethernet Backbone" by P. Ravasio, R. Marcogliese, and R. Novarese, pages 39-65, both in "Local Computer Networks" (edited by P. Ravasio, G. Hopkins, and N. Naffah; North Holland, 1982) and also in GB 2283252. The basic principles of such a scheme are that a first computer digitally samples a voice input signal at a regular rate (eg 8 kHz). A number of samples are then assembled into a data packet for transmission over the network to a second terminal, which then feeds the samples to a loudspeaker or equivalent device for playout, again at a constant 8 kHz rate. Voice transmission over the Internet is substantially similar to transmission over a LAN (which may indeed provide part of the Internet transmission path), but there tends to be less spare bandwidth available on the Internet. As a result, Internet phones normally compress the voice signal at the transmitting end, and then decompress it at the receiving end.
One of the major draw backs of the Internet telephony is that few of the Internet phones use the same standards. At present unless you have the same software as the person you are calling then it is impossible to connect with them. There are a variety of sound compression methods used in processing the voice signal and a large number use the Global Standard for mobile Communications (GSM) although some use proprietary compression algorithms. Even when vendors use the same compression algorithms there may be slight variations which can lead to compatibility problems. Another compatibility problem is that products tend to have different connection protocols and there seems little movement towards a common standard.
Another drawback of Internet telephony is that the quality of the voice signal is diminished by compressing and decompressing. In some circumstances a diminished quality signal can be so distorted as to make the message difficult to understand. Difficulty in understanding the message is further compounded if there is background noise, electrical interference, unclear speech or an unfamiliar accent or language.
A further drawback is that long delays can be introduced between messages from the participants which especially in the case of half-duplex transmission increases the difficulty in understanding the message. The factors which contribute to the delays include the time taken to compress and decompress the voice signal and transmission delays on the Internet when the network is busy and when the user has a slow modem connection.